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Communication Systems Laboratory Syllabus

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0% found this document useful (0 votes)
35 views102 pages

Communication Systems Laboratory Syllabus

Uploaded by

Vinotha
Copyright
© All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOC, PDF, TXT or read online on Scribd

EC3461- COMMUNICATION SYSTEMS LABORATORY,MECW

ANNA UNIVERSITY, CHENNAI


(R 2021)
EC3461 COMMUNICATIONSYSTEMSLABORATORY
SYLLABUS
1. AM-Modulator and Demodulator

2. FM-Modulator and Demodulator

3. Pre-Emphasis and De-Emphasis.

4. Signal sampling and TDM.

5. Pulse Code Modulation and Demodulation.

6. Pulse Amplitude Modulation and Demodulation.

7. Pulse Position Modulation and Demodulation and Pulse Width Modulation and Demodulation.

8. Digital Modulation–ASK, PSK, FSK.

9. Delta Modulation and Demodulation.

10. Simulation of ASK, FSK, and BPSK Generation and Detection


Schemes.

11. Simulation of DPSK, QPSK and QAM Generation and Detection Schemes.

12. Simulation of Linear Block and Cyclic Error Control coding Schemes.
EC3461- COMMUNICATION SYSTEMS LABORATORY,MECW

LIST OF EXPERIMENTS:

EXP NAME OF EXPERIMENS


NO.
CYCLE I

1. AM Modulator and Demodulator

2. FM Modulator and Demodulator

3. Pre-Emphasis and De-Emphasis.

4.(a) Signal Sampling and Reconstruction

4.(b) Time Division Multiplexing

5. Pulse Code Modulation and Demodulation

6. Pulse Amplitude Modulation and Demodulation.

7.(a) Pulse Position Modulation and Demodulation


7.(b) Pulse Width Modulation and Demodulation.

8.(a) Digital Modulation–ASK.

8.(b) Digital Modulation–FSK

8.(c) Digital Modulation–PSK.

9. Delta Modulation and Demodulation

CYCLE II

10.(a) Simulation of ASK generation Schemes

10.(b) Simulation of BPSK generation Schemes

10.(c) Simulation of FSK generation Schemes

11. Simulation of DPSK, QPSK and QAM Generation and Detection Schemes

12.(a) Simulation of Linear Block Coding

12.(b) Simulation of Cyclic Error Control Coding Schemes

BEYOND SYLLABUS
13 Communication Link Simulation

14 Line coding and Decoding Techniques


EC3461- COMMUNICATION SYSTEMS LABORATORY,MECW

INDEX

EXP. PAGE SIGNATURE


EXPERIMENTS MARKS
NO NO. OF THE STAFF

SIGNATURE OF THE STAFF MARK OUT OF--------------------


EC3461- COMMUNICATION SYSTEMS LABORATORY,MECW

CONNECTION DIGRAM
AM TRANSMITTER (MODULATOR & DEMODULTOR)
EC3461- COMMUNICATION SYSTEMS LABORATORY,MECW

[Link]
DATE:
AM MODULATOR AND DEMODULATOR

AIM
To transmit a modulating signal after amplitude modulation using AM transmitter and receive
the signal back after demodulating using AM receiver.
APPARATUS REQUIRED:

[Link]. Name of the Equipment/ Component Range Quantity


1. AM Transmitter kit - ACL 01 - 1
2. AM Receiver kit - ACL 01 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required
THEORY:
MODULATION THEORY:
Modulation is defined as the process by which some characteristics of a carrier signal is varied in
accordance with a modulating signal. The base band signal is referred to as the modulating signal and the
output of the modulation process is called as the modulation signal. The carrier frequency fc must be much
greater than the highest frequency components fm of the message signal m (t) i.e. fc>>fm. The modulation
index must be less than unity. If the modulation index is greater than unity, the carrier wave becomes over
modulated.
The modulating, carrier and modulated signals are given by
Vm(t) =Vmsinωmt ; VC(t) = VCsinωCt ; VAM(t) = VC (1+ma sinωmt) sinωCt The modulation
index is given by, ma = Vm / VC.
Vm = Vmax – Vmin and VC = Vmax + Vmin The amplitude of the modulated signal is given by,
Where Vm = maximum amplitude of modulating signal, VC = maximum amplitude of

carrier signal, Vmax = maximum variation of AM signal, Vmin = minimum variation of AM signal

DEMODULATION THEORY:
Demodulation is the reverse process of modulation. The detector circuit is employed
to separate the carrier wave and eliminate the side bands. Since the envelope of an AM wave has the
same shape as the message, independent of the carrier frequency and phase, demodulation can be
accomplished by extracting envelope. The depth of modulation at the detector output greater than unity
and circuit impedance is less than circuit load (Rl>Zm) results in clipping of negative peaks of
EC3461- COMMUNICATION SYSTEMS LABORATORY,MECW

modulating signal. It is called “negative clipping “.


TABULATION:

Amplitude Time Period


Parameter (V) in seconds

Message signal

Carrier signal

Modulated signal

Demodulated signal

Calculation of modulation index:


Practical calculation

Theoretical calculation

MODEL GRAPH:
PROCEDURE:

1. The circuit wiring is done as shown in diagram


2. A modulating signal input given to the Amplitude modulator
3. Now increase the amplitude of the modulating signal to the required level.
4. The amplitude and the time duration of the modulating signal are observed using CRO.
5. Finally the amplitude modulated output is observed from the output of amplitude modulator stage
and the amplitude and time duration of the AM wave are noteddown.
6. Calculate the modulation index by using the formula and verify them.
7. The final demodulated signal is viewed using CRO at the output of audio power amplifier stage.
Also the amplitude and time duration of the demodulated wave are noted down.

VIVA QUESTIONS:

1. What is the need of modulation and demodulation?


2. What is the range of frequency in commercial AM broadcasting?
3. What is analog modulation various techniques?
4. What is the difference between detector and demodulator?
5. Define the term Modulation index.
6. What are the main components of a RF receiver?
7. What is the Difference between Coherent and Non-coherent Demodulation?
8. Why the Intermediate Frequency Should is Carefully Chosen As?
9. What is the function of an Automatic Gain Control of the AM Receiver?
10. Define Amplitude Modulation?

RESULT:

Thus the AM signal was transmitted using AM trainer kit and the AM signal detected
using AM detector kit.
FREQUENCY DEVIATION f and MODULATION INDEX fm:

The frequency deviation f represents the maximum shift between the modulated signal frequency,
over and under the frequency of the carrier.

CONNECTION DIGRAM
FM MODULATOR

FM DEMODULATOR
[Link] :2
DATE:
FM MODULATION AND DEMODULATION

AIM:
To plot the modulation characteristics of FM modulator and demodulator and also to Observe and
measure frequency deviation and modulation index of FM.

APPARATUS REQUIRED:

[Link]. Name of the Equipment/ Component Range Quantity


1. FM Transmitter kit - ACL 03 - 1
2. FM Receiver kit - ACL 04 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required
THEORY:
Frequency modulation is a type of modulation in which the frequency of the high frequency
(carrier) is varied in accordance with the instantaneous value of the modulating signal.

FREQUENCY MODULATION GENERATION:


The circuits used to generate a frequency modulation must vary the frequency of a high
frequency signal (carrier) as function of the amplitude of a low frequency signal (modulating signal). In
practice there are two main methods used to generate FM.

DIRECT METHOD
An oscilloscope is used in which the reactance of one of the elements of the resonant circuit
depends on the modulating voltage. The most common device with variable reactance is the Varactor
or Varicap, which is a particular diode which capacity varies as function of the reverse bias voltage.
The frequency of the carrier is established with AFC circuits (Automated frequency control) or PLL
(Phase locked loop).

INDIRECT METHOD:
The FM is obtained in this case by a phase modulation, after the modulating signal has
been integrated. In this phase modulator the carrier can be generated by a quartz oscillator, and so its
frequency stabilization is easier. In the circuit used for the exercise, the frequency modulation is
generated by a Hartley oscillator, which frequency is determined by a fixed inductance and by
capacity (variable) supplied by varicap diodes.
TABULATION

Time Period
Parameter Amplitude (V)
in seconds

Message signal

Carrier signal

Modulated signal

Demodulated signal

MODEL GRAPH

We define modulation index m f the ratio between f and the modulating frequency f.
PROCEDURE:

1. Connect the power supply with proper polarity to the kit. While connecting this ensures that
the Power supply is OFF.
2. Switch on the power supply and carry out the following presetting as shownin circuit Diagram.
3. In the FM modulator set the level about 2Vpp and frequency knob to the minimumand switch on
1500 KHz.

4. Observe the FM modulated waveform from the RF/FM output of the FM modulator
measure frequency deviation and modulation index of FM.
5. For demodulation switch on the demodulator and carry out thefollowing demodulation connection
as shown in circuit diagram.
6. Observe the demodulated waveform and plot the graph.

VIVA QUESTION

1. Why frequency modulation is better than amplitude modulation?


2. What is the application of FM?
3. What is PLL?
4. What is FM modulation?
5. What is FM demodulation?
6. Define frequency deviation in FM?
7. Difference between narrowband and wideband FM?
8. What are the advantages of FM over AM?
9. What are the disadvantages of FM over AM?
10. Mention the types of digital modulation.

RESULT:
Thus the modulation characteristics of FM modulator and demodulator are observed and plotted.
BLOCK DIAGRAM OF PRE- EMPHASIS

BLOCK DIAGRAM OF DE- EMPHASIS

MODEL GRAPH:
PRE- EMPHASIS
[Link]
DATE:
PRE- EMPHASIS AND DE- EMPHASIS

AIM:
To obtain the frequency response of Pre-Emphasis and D- Emphasis.

APPARATUS REQUIRED:

[Link]. Name of the Equipment/ Component Range Quantity


1. Pre-Emphasis and D- Emphasis kit - 1
2. CRO 30 MHz 1
3. Function generator 1 MHZ 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required

THEORY
PRE- EMPHASIS
The higher the frequency modulation in FM, the greater the effect of noise. For higher
modulation frequency (fm), this effect can be reduced by increasing the value of the measurement factor
(mf). This can be done by increasing the difference Δf. δf can be increased by increasing the amplitude of
the modulation signal at a higher modulation frequency.

The process of optimizing the frequency components of a signal to improve the signal-to-noise
ratio. Therefore, the higher the amplitude of the modulation signal, the more we can improve its ability to
resist the noise of the modulation. This results in an improvement in signal-to-noise ratio, or SNR.
Artificially enhanced frequency change is called pre-emphasis. Preliminary analysis is performed on the
transmitter before frequency adjustment is made.

DE- EMPHASIS

The process applied at the end of receiving assistance or compensation for the first time for
artificial products that are frequently replaced is called highlighting. It is the process of high frequency
component of message signal to get back the original transmitted message signal. The artificially boosted
high frequencies signals are bought to their original amplitude using the de-emphasis circuit. After
demodulation, emphasis is made at the receiver.

The De-emphasis does exactly reverse of Pre-emphasis counterpart, it is used at the receiver part.
DE- EMPHASIS

TABULATION:

PRE EMPHASIS DE EMPHASIS


Frequency Input Output Gain= Frequency Input Output Gain=
in HZ Voltage Voltage 20 log in HZ Voltage Voltage 20 log
Vin Vout (Vin/Vout) Vin Vout (Vin/Vout)
PROCEDURE:
PRE- EMPHASIS
1. Give the connections as per the connection diagram.
2. Connect the circuit as per the circuit diagram .
3. Apply a sine wave to the input terminals of 2 VP-P (Vi)
4. By varying the input frequency in steps from 30HZ to 15KHZ with fixed amplitude, note down the output
amplitude (Vo) with respect to the input frequency.
5. Calculate the gain using the formula
Gain = 20 log (VO/ VI) db Where VO = output voltage in volts.
VI = Input voltage in volts.
Normalized gain (in dB) at any frequency - gain (in dB) at 500HZ.
6. Plot the frequency response.
DE-EMPHASIS:
1. Connect the output of the pre-emphasis circuit to the input of de-emphasis circuit.
2. Give the input from signal generator to the pre emphasis circuit.
3. Repeat steps 3,4 & 5 of Pre-Emphasis to de-emphasis also.

RESULT:
Thus the frequency response of Pre-Emphasis and D- Emphasis was obtained.
CONNECTION DIAGRAM

NATURAL SAMPLING

SAMPLE AND HOLD

FLAT – TOP SAMPLING


[Link](a)
DATE:
SIGNAL SAMPLING AND RECONSTRUCTION
AIM:
To sample and reconstruct the given signal using natural sampling, sample - hold and flat top
sampling techniques using DCL-01Kit.
APPARATUS REQUIRED:

[Link]. Name of the Equipment/ Component Range Quantity


1. DCL-01 trainer kit - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required

THEORY
Sampling is the process of splitting the given analog signal into different samples of equal
amplitudes with respect to time. There are two types of sampling namely natural sampling, flat top
sampling. Sampling should follow strictly the Nyquist Criterion i.e. the sampling frequency should be
twice higher than that of the highest frequency signal.
fs ≥ 2fm
Where,
fs= Minimum Nyquist Sampling rate (Hz),
fm= Maximum analog input frequency (Hz).
In natural sampling, the top of the sampled pulse follows the shape of the original signal. Since
the natural sampling increases the system complexity, the flat top sampling is mostly preferred in
practical case. The flat top sampling is achieved with the help of sample and hold circuit. Sample and
Hold circuits are used internally in analog to digital conversion. Here, the sampled signal obtained at each
sampling instant is hold until the next sampling instant. The sampling process must follow the sampling
theorem for proper signal reconstruction. In other words, the sampling frequency must be equal to twice
that of the highest frequency component present in the original signal.
The reconstructed signal is the succession of sine pulses weighted by x (nTs) these pulses are
interpolated with the help of a LPF. It is also called reconstruction filter or interpolation filter Natural
sampling is chopper sampling because the waveform of the sampled signal appears to be chopped off
from the original signal waveform. The top of the samples remains constant and equal to instantaneous

value of x(t) at start of sampling fs = 1/Ts.

TABULATION
NATURAL SAMPLING

Amplitude Time Period in


Parameters
in Volts Seconds
Input Signal

Clock Signal

Natural sampling Output

Filter Output

FLAT TOP SAMPLING

Amplitude in Time Period in


Parameters
Volts Seconds
Input Signal

Clock Signal

Flat top Output

Filter Output

MODEL GRAPH

Duty cycle calculation: D = Ton / (Ton + Toff) = %

PROCEDURE:
7. Give the connections as per the connection diagram.
8. Put the duty cycle selector switch in position 50%.
9. Connect the modulating signal of 1 KHz frequency to BUFIN and measure its amplitude and
time period.
10. Connect the sampling frequency clock in the internal mode INT CLK using the sampling signal
selector switch SW4.
11. Set the sampling frequency to 8 KHz and note down the amplitude and time period of the
sampling signal.
12. Observe the natural sampling output at OUT terminal and note down the amplitude and time
period of the sampled signal.
13. Give the sampled signal to the 2nd order low pass filter circuit and observe the reconstructed
signal.
14. Note down the amplitude and time period of the reconstructed signal.
15. Repeat the same procedure for flat top sampling, Sample and hold and note down the readings.
16. Plot the readings in the graph.

VIVA QUESTIONS:

1. What is the need of sampling?


2. What is sampling rate?
3. What is Nyquist rate of sampling?
4. Mention the types of sampling.
5. State the sampling theorem.
6. What is natural sampling?
7. Differentiate three types of sampling techniques.
8. What is the function sample & hold circuit?
9. What is aliasing effect and how it can be eliminated?
10. State the role of duty cycle in sampling circuit.
11. What is the function reconstruction filter?
12. What is the use of sampling and reconstruction technique in communication engineering?

RESULT:
Thus the given signal is sampled and reconstructed for natural sampling, sample and hold and
flat top sampling also plotted the observed waveforms.
CONNECTION DIAGRAM
[Link](b)
DATE:
TIME DIVISION MULTIPLEXING

AIM:

To perform four channel Time Division multiplexing and De multiplexing using DCL – 02 trainer kit.

APPARATUS REQUIRED:

[Link]. Name of the Equipment/ Component Range Quantity


1. DCL-02 trainer kit - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required

THEORY:

Time Division Multiplexing (TDM) is a technique of transmitting different source signals on the same
channel at different time slots. That is several information can be transmitted over a single channel by
sending samples from different information sources at different moments. TDM is widely used in digital
communication systems to increase the efficiency of the transmitting medium. TDM can be achieved by
electronically switching the samples such that they interleave sequentially at correct instant in time without
mutual interference. A major problem in any TDM system is the synchronization of the transmitter and
receiver timing circuits. The transmitter and receiver must switch at the same time and frequency. TDM
based on analog modulation, the time slots are separated by guard slots to prevent crosstalk between the
channels.
In PAM, PPM the pulse is present for a short duration and for most of the time between the two pulses
no signal is present. This free space between the pulses can be occupied by pulses from other channels.
Thus, time division multiplexing makes maximum utilization of the transmission channel. Each channel to
be transmitted is passed through the low pass filter. The outputs of the low pass filters are connected to the
rotating sampling switch (or) commutator.
It takes the sample from each channel per revolution and rotates at the rate of f s. Thus the sampling
frequency becomes fs the single signal composed due to multiplexing of input channels. These channels
signals are then passed through low pass reconstruction filters. If the highest signal frequency present in all
the channels is fm, then by sampling theorem, the sampling frequency fs must be such that fs≥2fm.
Therefore, the time space between successive samples from any one input will be Ts=1/fs, and Ts≤ 1/2fm.
TABULATION:

Amplitude Time Period


Parameters
in Volts in Seconds
Message Signal 1

Message Signal 2

Multiplexed Ouput

Demodulated Output 1

Demodulated Output 2
PROCEDURE:

1. Give the connections as per the connection diagram.


2. Connect the power supply with proper polarity to the kit DCL-02 and switch it ON.
3. Connect 250 Hz, 500 Hz, 1 KHz, 2 KHz sine wave signal from the function generator to
the multiplexer input channel CH0, CH1, CH2 and CH3 by means of connecting cords.
4. Connect the multiplexer OUT TXD to the transmitter section to the de-multiplexer input
RXD of the receiver section.
5. Connect the output of the receiver section CH0, CH1, CH2 and CH3 to the IN0, IN1,
IN2 and IN3 of the filter section.
6. Connect the sampling clock TXCLK and the channel identification clock SYNC of
the receiver section respectively.
7. Set the amplitude of the input sine wave as desired.
8. Take the observations and draw the required graphs.

VIVA QUESTIONS:

1. What is meant by multiplexing technique?


2. What are the two types of multiplexing?
3. When do you prefer TDM to FDM?
4. When would you prefer FDM to TDM?
5. Differentiate TDM and FDM.
6. Define crosstalk effect in PAM/TDM system.
7. What is the transmission bandwidth of a PAM/TDM channel?
8. State the advantages of TDM.
9. State the disadvantages of TDM.
10. Why are the bands limiting filters used at the PAM-TDM transmitter?
11. In TDM, multiple signals share a channel by transmitting in different ……..
12. The minimum sampling rate for a voice channel is………..

RESULT:

Thus the four channel Time division multiplexing and de-multiplexing using DCL-02 kit was
performed and obtained its waveform.
CONNECTION DIAGRAM:

MODEL GRAPH:
[Link] :
5 DATE:
PULSE CODE MODULATION AND DEMODULATION

AIM
To generate a PCM signal using PCM modulator and detect the message signal from PCM signal by using
PCM demodulator.

APPARATUS REQUIRED:

[Link]. Name of the Equipment/ Component Range Quantity


1. PCM Trainer kit - DCL 03, DCL 04 - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required
THEORY:
Pulse code modulation is known as digital pulse modulation technique. It is the process
in which the message signal is sampled and the amplitude of each sample is rounded off to the nearest
one of the finite set of allowable values. It consists of three main parts transmitter, transmitter path and
receiver. The essential operation in the transmitter of a PCM system are sampling, quantizing and
encoding. The band pass filter limits the frequency of the analog input signal. The sample and hold
circuit periodically samples the analog input signal and converts those to a multi-level PAM signal.
The ADC converts PAM samples to parallel PCM codes which are converted to serial binary data in
parallel to serial converter and then outputted on the transmission line as serial digital pulse. The
transmission line repeaters are placed at prescribed distance to regenerate the digital pulse.
In the receiver serial to parallel converter converts serial pulse received from the
transmission line to parallel PCM codes. The DAC converts the parallel PCM codes to multi- level
PAM signals. The hold circuit is basically a Low Pass Filter that converts the PAM signal back to its
original analog form.

ADVANTAGES:
1. Secrecy
2. Noise resistant and hence free from channel interference
DISADVANTAGE:
1. Requires more bandwidth
TABULATION
TRANSMITTER:

Time period in mSec


Signal Amplitude in V
Message Signal (500Hz)
Message Signal (1KHz)
OUT 0
OUT 2
Multiplexer OUT

RECEIVER:

Time period in mSec


Signal Amplitude in V
DAC OUT
CH 0
CH 1
OUT 0
OUT 1

PROCEDURE:
1. Refer the block diagram and carry out the following connections and switch settings.
2. Connect the power supply with power polarity to the kit and switch it „ON‟.
3. Put the switch sw1 to „FAST‟ mode.
4. Select 500Hz and 1 KHz sine wave signals generated on board.
5. Connect the signals to „CH0‟ and „CH1‟ of sample and hold circuit.
6. The output of sample and hold circuit „OUT0‟ and „OUT1‟ are given to the multiplexer, then to the pulse
code modulation logic.
7. The output „TXDATA‟ is connected to „RXDATA‟ of pulse code demodulation logic.
8. The output „DAC OUT‟ is given to „IN‟ of „DEMUX‟.
9. Then the output of „DEMUX‟ –„OUT0‟ and „OUT1‟ are given to „IN0‟ and „IN1‟ of filter.
10. Observe the pulse code demodulated output at „OUT0‟ and „OUT1‟.

VIVA QUESTIONS:

1. What is PCM?
2. How bits are needed to encode N different levels?
3. Define step size?
4. How to calculate Step size in PCM?
5. Define Quantization error.
6. What is the max value of Quantization error?
7. What are the applications of PCM?
8. What are the disadvantages of Pulse code modulation?
9. Define Differential pulse code modulation?
[Link] DPCM is better than PCM?

RESULT

Thus the PCM signal was generated using PCM modulator and the message signal was
detected from PCM signal by using PCM demodulator.

PULSE AMPLITUDE MODULATOR

Audio Frequency
Signal Generator Monostable
Multivibrator PAM O/P

Astable Multivibrator

DEMODULATOR

Demodulated
PAM Low Pass
modulated Signal Demodulated
Amplifier
Signal Signal

TABULATION:

Signal Amplitude(V) Time(ms)


[Link] PULSE AMPLITUDE MODULATION AND DEMODULATION
Date:
AIM:
To obtain the pulse amplitude modulated and demodulated waveform for given message signal.
APPARATUS REQUIRED:
1. PAM Modulation Kit and Demodulation Kit
2. CRO
3. Connecting probes
THEORY:
The amplitude of the train of rectangular pulses are varied in accordance with the message signal is
called pulse amplitude modulation.
The message signal is an analog signal
PAM use sampling of two types
i) Natural sampling
ii) Flat top sampling
It is better to use flat top PAM, because during transmission noise interferes the top of pulses. This noise can
be removed easily.
Band width of the transmission channel depends on width of the pulse.
In PAM, noise interference is high.
It requires less transmitting power.
Multiplexing of several PAM signal at different time is possible with PAM.

PROCEDURE:-

1. Connections must be given as per the diagram.


2. Low frequency message signal is given as one input to PAM modulator.
3. Carrier pulse signal is given as another input to PAM modulator.
4. The pulse amplitude modulated waveform obtained is viewed in CRO.
5. Readings are taken for message, carrier and pulse amplitude modulated wave.
6. The modulated wave is given as input to demodulator
7. The demodulated output is noted in CRO.
8. The graph is plotted for PAM.

V
Message Signal

Time (t)

Train of Rectangular Pulse(Sampling pulse)

Time (t)

Amplitude
(Volt)
Pulse Amplitude
Modulated wave

Time (t)

Demodulated Signal

Time (t)
RESULT:

Thus for given message signal PAM modulation and demodulation is obtained. It is plotted
in graph

BLOCKDIAGRAM

Audio Frequency
Signal Generator

Pulse Pulse Position Modulated


Position Output
Astable Modulator
Multivibrator
Pulse Position
Low
Modulated Demodulated
Pass Amplifier
Signal Signal
Filter

TABULATION:

Signal Amplitude(V) Time(ms)


[Link](a) PULSE POSITION MODULATION AND DEMODULATION
Date:

AIM:
To obtain the pulse Position modulated and demodulated waveform for given message signal.
APPARATUS REQUIRED:
4. PPM Modulation Kit and Demodulation Kit
5. CRO
6. Connecting probes
THEORY:
The positions of the regularly spaced train of pulses are varied in accordance with the message signal.
Bandwidth of transmission channel depends on rising time of the pulse.
BT≥1/2tr
PPM has minimum noise interference. This modulation is similar to PM. It requires less transmitting power.
PT(PPM) <<PT (PWM).

PROCEDURE:-

1. Connections must be given as per the diagram.


2. Low frequency message signal is given as one input to PPM modulator.
3. Carrier pulse signal is given as another input to PPM modulator.
4. The pulse amplitude modulated waveform obtained is viewed in CRO.
5. Readings are taken for message, carrier and pulse amplitude modulated wave.
6. The modulated wave is given as input to demodulator
7. The demodulated output is noted in CRO.
8. The graph is plotted for PPM.

RESULT:
Thus for given message signal PPM modulation and demodulation is obtained. It is plotted in graph
BLOCKDIAGRAM

AudioFrequency
SignalGenerator

Pulse PWM Output


Width
Astable Modulator
Multivibrator
Pulse Width
Low
Modulated Demodulated
Pass Amplifier
Signal Signal
Filter
TABULATION:

Signal Amplitude(V) Time(ms)


[Link](b) PULSE WIDTH MODULATION AND DEMODUKATION
Date:

AIM:
To obtain the pulse Width modulated and demodulated waveform for given message signal.
APPARATUS REQUIRED:
1. PWM Modulation Kit and Demodulation Kit
2. CRO
3. Connecting probes
THEORY:
The widths of regularly spaced train of pulses are varied in accordance with the message signal.
Bandwidth of transmission channel depends on rise time of the pulse.
BT>>1/2tr
PWM has less noise interference. This type of modulation is similar to FM.
Transmitting power PT (PWM) >> PT (PPM).

PROCEDURE:-

1. Connections must be given as per the diagram.


2. Low frequency message signal is given as one input to PWM modulator.
3. Carrier pulse signal is given as another input to PWM modulator.
4. The pulse amplitude modulated waveform obtained is viewed in CRO.
5. Readings are taken for message, carrier and pulse amplitude modulated wave.
6. The modulated wave is given as input to demodulator
7. The demodulated output is noted in CRO.
8. The graph is plotted for PWM.
MODELGRAPH:-

MessageSignal

time (t)

Sampling pulse
(Train of
rectangularpulses)

time(t)

Pulse width
modulatedwave
Amplitude
(volt)
time(t)

Demodulatedwave

time(t)
RESULT:
Thus for given message signal PWM modulation and demodulation is obtained. It is plotted in graph

BLOCKDIAGRAM

ASKMODULATOR

ASKDEMODULATOR
ExNo:8(a) GENERATION AND DETECTION OF ASK
Date:

AIM:

To construct and generate Amplitude Shift Keying signal and detect the message signal.

APPARATUS REQUIRED:

1. ASK kit,
2. CRO
3. Connecting probes

THEORY:

ASK or ON-OFF key is the simplest digital modulation technique. In this method there is
only one unit energy carrier it is switched ON/OFF depending upon the input binary sequence to
transmit symbol 0 & 1. No pulse is transmitted output contains some complete no of cycle of
carrier frequency. The disadvantage of ASK is the modulated carrier signal is not continuously
transmitted. The peak power requirement is also high. The bit error probability rate is also not
required in this technique.

PROCEDURE

1. Make connections as shown in the diagram.


2. Set the input signal and carrier signal.
3. Obtain ASK signal
4. Measure the amplitude and frequency
5. Obtain the demodulated output.
MODELGRAPHH

TABULARCOLUMN

[Link] Nameofthesignal AmplitudeinV TimeperiodinSec FrequencyinHz

1 ModulatingSignal

2 CarrierSignal

3 ModulatedSignal

4 Demodulated
Signal
RESULT:

Thus the Amplitude Shift Keying signal was generated and the message signal was
reconstructed.

37
BLOCK DIAGRAM

FSK MODULATOR

FSK DEMODULATOR:

38
ExpNo:8(b) GENERATION AND DETECTION OF FSK
Date:

AIM

To generate a Frequency Shift Keying signal using FSK modulator and detect the
message signal from FSK signal using FSK detector.

APPARATUS REQUIRED:
[Link] kit,

[Link]

[Link] probes

THEORY

Frequency Shift Keying is the process generating a modulated signal from a digital data
input. If the in coming bit is 1, a signal with frequency f1is sent for the duration of the bit. If the
bit is 0, a signal with frequency f2 is sent for the duration of this bit. This is the basic principle
behind FSK modulation.
Basically a 555 timer is used as an Astable multivibrator, which generates a clock pulse of
frequency determined by the values of R and C in this circuit.Thisisdividedby2,4,8 and 16using
74163IC,andtwo of these outputs are used in a NAND logic gates circuit, to generate a FSK
modulated wave. To this NAND gates circuit a binary data sequence is also supplied. The circuit
operation causes a frequencyf1 for bit 1, and f2 for bit 0.
In the demodulator circuit, the FSK modulated signal is applied to a high Q tuned filter.
This filter is tuned to the frequency of either 0 or1. This filter passes the selected frequency and
rejects the other. The output is then passed through a FWR (Full Wave Rectifier) circuit and the
output is now above zero volts only. It is then passed through a comparator; if the input to the
comparator is greater than threshold value, the output is 1, else it is 0. This digital output of the
comparator is the demodulated FSK output.
PROCEDURE:

1. Make connections as shown in the diagram.


2. Set the input signal and carrier signal.
3. Obtain FSK signal
4. Measure the amplitude and frequency
5. Obtain the demodulated output.
39
MODELGRAPH

41
TABULAR COLUMN

[Link] Name of the signal Amplitude in V Time period in Sec Frequency in Hz

1 Modulating Signal

2 CarrierSignal1

3 CarrierSignal2

4 Modulated Signal

5 Demodulated
Signal

RESULT

Thus the Frequency Shift Keying signal was generated using FSK modulator and the
message signal was detected from FSK signal using FSK detector.

42
BPSK MODULATOR

BPSK DEMODULATOR

43
Ex:No:8(c) GENERATION AND DETECTION OF PSK
Date:

AIM:To construct and generate Phase Shift Keying signal and detect the message signal.

APPARATUSREQUIRED

1. PSK kit

2. CRO

3. Connecting probes

THEORY

PSK is a digital modulation scheme which is analogues to phase modulation. In binary


phase shift keying two output phases are possible for a single carrier frequency one out of phase
represent logic 1 and logic 0. As the input digital binary signal change state the phase of output
carrier shift two angles that are 180o out of phase.
In a PSK modulator the carrier input signal is multiplied by the digital data. The input
carrier is multiplied by either a positives or negatives consequently the output signal is either
+1sinwct or - 1sinwct. The first represent a signal that is phase with the reference oscillator the
latter a signal that is 180o out of phase with the reference oscillator. Each time a change in input
logic condition will change the output phase consequently for PSK the output rate of change equal
to the input rate range and widest output bandwidth occurs when the input binary data are
alternating 1/0 sequence. The fundamental frequency of an alternate 1/0 bit sequence is equal to
one half of the bit rate.

PROCEDURE

1. Make connections as shown in the diagram.


2. Set the input signal and carrier signal.
3. Obtain PSK signal
4. Measure the amplitude and frequency
5. Obtain the demodulated output.

43
MODELGRAPH
TABULAR COLUMN:

[Link] Name of the signal Amplitude in V Time period in Sec Frequency in Hz

1 Modulating Signal

2 Carrier Signal

3 Modulated Signal

4 Demodulated
Signal
RESULT

Thus the Phase Shift Keying signal was generated using PSK modulator and
the message signal was detected.

CONNECTION DIAGRAM
DELTA MODULATION AND DEMODULATION
[Link] : 9
DATE :
DELTA MODULATION AND DEMODULATION
AIM
To transmit an analog message signal in its digital form and again reconstruct back the original
analog message signal at receiver by using Delta Modulator and Demodulator.

APPARATUS REQUIRED:

[Link] Name of the Equipment/ Component Range Quantity


Delta modulation/ demodulation
1. - 1
trainer kit, DCL 07
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required
THEORY:
Delta modulation uses a single bit PCM code to achieve digital transmission of analog
signal. With conventional PCM, each code is a binary representative of both the sign and magnitude
of a particular sample. The algorithm of delta modulation is simple if the current sample is smaller
than the previous sample a logic0 is transmitted. If the current sample is larger than the previous
sample logic 1 is transmitted.

ADVANTAGES:
1. Simple system/circuitry - Cheap
2. Single bit encoding allows us to increase the sampling rate or to transmit
more information at some sampling rate for the given system BW.

DISADVANTAGES:
1. Noise and distortion.
2. Major drawback is that it is unable to pass DC information.
APPLICATIONS:
Digital voice storage, Voice transmission, Radio communication devices such TV remotes.
Adaptive delta modulation is delta modulation system where the step size of DAC is
automatically varied, depending on the amplitude characteristics of the analog input signal.
MODEL GRAPH

TABULATION

Time Period
Parameter Amplitude (V)
in seconds
Input signal

Integrator 1 output

Sampler output

Integrator 1 output

Filter output

Demodulated output
PROCEDURE:
1. Refer to the block diagram and carry out the following connections
2. Connect the power supply with proper polarity to the kit DCL-07 and switch it ON.
3. Select the sine wave input 250Hz or through port P1 and connect port 250Hz to port
IN of input buffer.
4. Connect the output of buffer „OUT‟ to digital samples input port „IN1‟.
5. Then select clock rate of 8 KHz by pressing switch S1. Selected clock is indicated
by LED glow.
6. Keep switch S2 in (∆) deltaposition.
7. Connect output of a digital sampler port „OUT‟ to input port „IN‟ of integrator.
8. Connect output of integrator 1 port „OUT‟ to input port „IN2‟ of digital sampler.
9. The digital sampler „OUT‟ is given to the input of the output buffer.
10. The output of the buffer is given to the second order input of the fourth
order Butterworth filter.

VIVA QUESTIONS:
1. Define Delta modulation? Why it is better?
2. What is granular noise? Define slope overload?
3. When granular noise and slope overload occur in Delta modulation?
4. What is Adaptive Delta Modulation and what are the advantages?
5. Compare all Digital pulse modulation techniques (PCM, DPCM, DM and ADM)
6. In digital transmission, the modulation technique that requires minimum bandwidth is
…………
7. In Delta Modulation, the bit rate is ………….
8. What are the applications of DM?
9. Mention the advantage of DM?
10. What is temporal waveform coding?

RESULT:
Thus the analog message signal in its digital form was transmitted and again the original analog message
signal was reconstructed at receiver by using Delta modulator and Demodulator.
[Link] : 10(a)
DATE:
SIMULATION OF ASK GENERATION AND DETECTION SCHEME
AIM
To generate and demodulate amplitude shift keyed (ASK) signal using MATLAB.
THEORY
Generation of ASK
Amplitude shift keying - ASK - is a modulation process, which imparts to a sinusoid two or
more discrete amplitude levels. These are related to the number of levels adopted by the digital
message. For a binary message sequence there are two levels, one of which is typically zero. The data
rate is a sub-multiple of the carrier frequency. Thus the modulated waveform consists of bursts of a
sinusoid. One of the disadvantages of ASK, compared with FSK and PSK, for example, is that it has
not got a constant envelope. This makes its processing (eg, power amplification) more difficult, since
linearity becomes an important factor. However, it does make for ease of demodulation with an
envelope detector.

Demodulation
ASK signal has a well defined envelope. Thus it is amenable to demodulation by an
envelope detector. Some sort of decision-making circuitry is necessary for detecting the message. The
signal is recovered by using a correlator and decision making circuitry is used to recover the binary
sequence.
Algorithm
Initialization commands
ASK modulation
1. Generate carrier signal.
2. Start FOR loop
3. Generate binary data, message signal(on-off form)
4. Generate ASK modulated signal.
5. Plot message signal and ASK modulated signal.
6. End FOR loop.
7. Plot the binary data and carrier.
ASK demodulation
1. Start FOR loop
2. Perform correlation of ASK signal with carrier to get decision variable
3. Make decision to get demodulated binary data. If x>0, choose „1‟ else choose „0‟
4. Plot the demodulated binary data.
Program
%ASK Modulation
clc;
clear
all;
close
all;
%GENERATE CARRIER
SIGNAL Tb=1; fc=10;
t=0:Tb/100:1;
c=sqrt(2/Tb)*sin(2*pi*fc*t);
%generate message
signal N=8;
m=rand(1,N)
; t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2]
if
m(i)>0.5
m(i)=1;
m_s=ones(1,length(t)
); else
m(i)=0;
m_s=zeros(1,length(t));
end
message(i,:)=m_s;
%product of carrier and
message ask_sig(i,:)=c.*m_s;
t1=t1+(Tb+.01); t2=t2+
(Tb+.01);
%plot the message and ASK signal
subplot(5,1,2);axis([0 N -2
2]);plot(t,message(i,:),'r');
title('message signal');xlabel('t--->');ylabel('m(t)');grid
on hold on
subplot(5,1,4);plot(t,ask_sig(i,:));
title('ASK signal');xlabel('t--->');ylabel('s(t)');grid
on hold on

end
hold
off
%Plot the carrier signal and input binary
data subplot(5,1,3);plot(t,c);
title('carrier signal');xlabel('t--->');ylabel('c(t)');grid on
subplot(5,1,1);stem(m);
title('binary data bits');xlabel('n--->');ylabel('b(n)');grid on
% ASK Demodulation
t1=0;t2=T
b for
i=1:N
t=[t1:Tb/100:t2]
%correlator
x=sum(c.*ask_sig(i,:));
%decision
device if x>0
demod(i)=1;
else
demod(i)=0;
end t1=t1+
(Tb+.01);
t2=t2+
(Tb+.01);
end
%plot demodulated binary data
bits subplot(5,1,5);stem(demod);
title('ASK demodulated signal'); xlabel('n--->');ylabel('b(n)');grid on
SIMULATION WAVEFORM
RESULT
The program for ASK modulation and demodulation has been simulated in MATLAB and necessary graphs
are plotted.
[Link](b)
DATE: SIMULATION OF BPSK GENERATION AND DETECTION SCHEMES
AIM
To generate and demodulate Binary phase shift keyed (BPSK) signal using MATLAB.
Generation of PSK signal

PSK is a digital modulation scheme that conveys data by changing, or modulating, the phase of
a reference signal (the carrier wave). PSK uses a finite number of phases, each assigned a unique pattern
of binary digits. Usually, each phase encodes an equal number of bits. Each pattern of bits forms the
symbol that is represented by the particular phase. The demodulator, which is designed specifically for
the symbol- set used by the modulator, determines the phase of the received signal and maps it back to
the symbol it represents, thus recovering the original data.

In a coherent binary PSK system, the pair of signal S1(t) and S2 (t) used to represent
binary symbols 1 & 0 are defined by
S1 (t) = √2Eb/ Tb Cos 2πfct
S2 (t) =√2Eb/Tb (2πfct+π) = - √ 2Eb/Tb Cos 2πfct where 0 ≤ t< Tb and
Eb = Transmitted signed energy for bit
The carrier frequency fc =n/Tb for some fixed integer n.
Antipodal Signal:
The pair of sinusoidal waves that differ only in a relative phase shift of 180° are called antipodal
signals.
BPSK Transmitter

Program
% BPSK modulation
clc;
clear
all;
close
all;
%GENERATE CARRIER
SIGNAL Tb=1;
t=0:Tb/100:Tb;
fc=2;
c=sqrt(2/Tb)*sin(2*pi*fc*t
);
%generate message
signal N=8;
m=rand(1,N)
; t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2]
if
m(i)>0.5
m(i)=1;
m_s=ones(1,length(t)
); else
m(i)=0;
m_s=-
1*ones(1,length(t)); end
message(i,:)=m_s;
%product of carrier and message
signal bpsk_sig(i,:)=c.*m_s;
%Plot the message and BPSK modulated signal
subplot(5,1,2);axis([0 N -2
2]);plot(t,message(i,:),'r');
title('message signal(POLAR form)');xlabel('t---
>');ylabel('m(t)'); grid on; hold on;
subplot(5,1,4);plot(t,bpsk_sig(i,:));
title('BPSK signal');xlabel('t---
>');ylabel('s(t)');

grid on; hold on;

The input binary symbols are represented in polar form with symbols 1 & 0 represented by
constant amplitude levels √Eb& -√Eb. This binary wave is multiplied by a sinusoidal carrier in a
product modulator. The result in a BSPK signal.
BSPK Receiver:

The received BPSK signal is applied to a correlator which is also supplied with a locally generated
reference signal c1 (t). The correlated o/p is compared with a threshold of zero volts. If x> 0, the receiver
decides in favour of symbol 1. If x< 0, it decides in favour of symbol 0.
Algorithm
Initialization commands
BPSK modulation
8. Generate carrier signal.
9. Start FOR loop
10. Generate binary data, message signal in polar form
11. Generate PSK modulated signal.
12. Plot message signal and PSK modulated signal.
13. End FOR loop.
14. Plot the binary data and carrier.
BPSK demodulation
1. Start FOR loop
2. Perform correlation of PSK signal with carrier to get decision variable
3. Make decision to get demodulated binary data. If x>0, choose „1‟ else choose „0‟
4. Plot the demodulated binary data.

t1=t1+1.01; t2=t2+1.01;
end
hold
off
%plot the input binary data and carrier
signal subplot(5,1,1);stem(m);
title('binary data bits');xlabel('n---
>');ylabel('b(n)'); grid on;
subplot(5,1,3);plot(t,c);
title('carrier signal');xlabel('t---
>');ylabel('c(t)'); grid on;
% PSK Demodulation
t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2
]
%correlator
x=sum(c.*bpsk_sig(i,:));
%decision device
if x>0 demod(i)=1; else demod(i)=0; end t1=t1+1.01; t2=t2+1.01;
end
%plot the demodulated data
bits
subplot(5,1,5);stem(demod);
title('demodulated data');xlabel('n---
>');ylabel('b(n)'); grid on;
SIMULATION WAVEFORM

RESULT
Thus the program for BPSK modulation and demodulation has been simulated in
MATLAB and necessary graphs are plotted.
[Link] : 10(c)
DATE:
SIMULATION OF FSK GENERATION AND DETECTION SCHEME
AIM

To generate and demodulate frequency shift keyed (FSK) signal using MATLAB
THEORY
Generation of FSK
Frequency-shift keying (FSK) is a frequency modulation scheme in which digital
information is transmitted through discrete frequency changes of a carrier wave. The simplest
FSK is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit binary (0s and
1s) information. With this scheme, the "1" is called the mark frequency and the "0" is called the
space frequency.
In binary FSK system, symbol 1 & 0 are distinguished from each other by transmitting one of
the two sinusoidal waves that differ in frequency by a fixed amount.
Si (t) = √2E/Tb cos 2πf1t 0≤ t ≤Tb 0 elsewhere
Where i=1, 2 &Eb=Transmitted
energy/bit
Transmitted freq= ƒi = (nc+i)/Tb, and n = constant (integer), Tb = bit interval
Symbol 1 is represented by S1 (t)
Symbol 0 is represented by S0 (t)
BFSK Transmitter

The input binary sequence is represented in its ON-OFF form, with symbol 1
represented by constant amplitude of √Ebwith & symbol 0 represented by zero volts. By using
inverter in the lower channel, we in effect make sure that when symbol 1is at the input, The
two frequency f1& f2 are chosen to be equal integer multiples of the bit rate 1/[Link]
summing the upper & lower channel outputs, we get BFSK signal.
Program
% FSK Modulation
clc;
clear
all;
close
all;
%GENERATE CARRIER
SIGNAL Tb=1; fc1=2;fc2=5; t=0:
(Tb/100):Tb;
c1=sqrt(2/
Tb)*sin(2*pi*fc1*t);
c2=sqrt(2/Tb)*sin(2*pi*fc2*t
);
%generate message
signal N=8;
m=rand(1,N)
; t1=0;t2=Tb
for i=1:N
t=[t1:(Tb/100):t2]
if
m(i)>0.5
m(i)=1;
m_s=ones(1,length(t));
invm_s=zeros(1,length(t))
; else
m(i)=0;
m_s=zeros(1,length(t));
invm_s=ones(1,length(t)
); end
message(i,:)=m_s;
%Multiplier
fsk_sig1(i,:)=c1.*m_s;
fsk_sig2(i,:)=c2.*invm_
s;
fsk=fsk_sig1+fsk_sig2;
%plotting the message signal and the modulated
signal subplot(3,2,2);axis([0 N -2
2]);plot(t,message(i,:),'r');

title('message signal');xlabel('t --- >');ylabel('m(t)');grid on;hold on;


BFSK Receiver

The receiver consists of two correlators with common inputs which are supplied with
locally generated coherent reference signals c1(t) and c2 (t).
The correlator outputs are then subtracted one from the other, and the resulting
difference x is compared with a threshold of zero volts. If x >0, the receiver decides in favour
of symbol 1 and if x <0, the receiver decides in favour of symbol 0.
Algorithm
Initialization
commands FSK
modulation
1. Generate two carriers signal.
2. Start FOR loop
3. Generate binary data, message signal and inverted message signal
4. Multiply carrier 1 with message signal and carrier 2 with inverted message signal
5. Perform addition to get the FSK modulated signal
6. Plot message signal and FSK modulated signal.
7. End FOR loop.
8. Plot the binary data and carriers.

FSK demodulation
1. Start FOR loop
2. Perform correlation of FSK modulated signal with carrier 1 and carrier 2 to get two
decision variables x1 and x2.
3. Make decisionon x = x1-x2 to get demodulated binary data. If x>0, choose „1‟ else choose „0‟.
4. Plot the demodulated binary data.

subplot(3,2,5);plot(t,fsk(i,:));
title('FSK signal');xlabel('t--- >');ylabel('s(t)');grid on;hold on; t1=t1+
(Tb+.01); t2=t2+(Tb+.01);
end
hold
off
%Plotting binary data bits and carrier
signal subplot(3,2,1);stem(m);
title('binary data');xlabel('n --- >'); ylabel('b(n)');grid
on; subplot(3,2,3);plot(t,c1);
title('carrier signal-1');xlabel('t --- >');ylabel('c1(t)');grid
on; subplot(3,2,4);plot(t,c2);
title('carrier signal-2');xlabel('t --- >');ylabel('c2(t)');grid on;

% FSK Demodulation
t1=0;t2=T
b for
i=1:N
t=[t1:(Tb/100):t2]
%correlator
x1=sum(c1.*fsk_sig1(i,:)
);
x2=sum(c2.*fsk_sig2(i,:)
); x=x1-x2;
%decision
device if x>0
demod(i)=1;
else
demod(i)=0;
end t1=t1+
(Tb+.01);
t2=t2+
(Tb+.01);
end
%Plotting the demodulated data
bits subplot(3,2,6);stem(demod);
title(' demodulated data');xlabel('n--- >');ylabel('b(n)'); grid on;
SIMULATION WAVEFORM

RESULT
Thus the program for FSK modulation and demodulation has been simulated in
MATLAB and necessary graphs are plotted.
[Link] : 11
DATE:
SIMULATION OF DPSK, QPSK AND QAM GENERATION AND DETECTION
SCHEMES
AIM
To simulate DPSK , QPSK and QAM Generation and Detection Schemes using MATLAB.

SOFTWARE REQUIRED:
MATLAB
PROGRAM FOR DPSK:
clc;
clear
all;
rng default
M = 6; % Alphabet size
dataIn = randi([0 M-1],1011,1); % Random
message txSig = dpskmod(dataIn,M); % Modulate
rxSig =
txSig*exp(2i*pi*rand());
dataOut =
dpskdemod(rxSig,M); errs =
symerr(dataIn,dataOut)
errs =
symerr(dataIn(2:end),dataIn(2:end))
figure
subplot(2,2,1)
plot(dataIn)
title('DATA')
subplot(2,2,2)
plot(txSig)
title('DPSK
SIGNAL')
subplot(2,2,3)
plot(rxSig)
title('Received
DPSK')
subplot(2,2,4)
plot(dataOut)
title('DATA RECEIVED')
SIMULATION WAVEFORM

PROGRAM FOR QPSK:

QPSK
clc; clear all;
t=0:0.0001:0.25;
m=square(2*pi*10*t);
c1=sin(2*pi*60*t);
c2=sin(2*pi*60*t+180);
for i=1:2500
if(mod(i,1000))<500
s(i)=c1(i);
else
s(i)=-c2(i);
end
end
subplot(4,1,1);
plot(t,m,'k','linewidth',5);
title('polor representation of message 1 0 1 0 1
0'); xlabel('time'); ylabel('amplitude')
subplot(4,1,2); plot(c1);
title('frequency 1');
xlabel('time'); ylabel('amplitude');
subplot(4,1,3); plot(c2);
title('frequency 2');
xlabel('time'); ylabel('amplitude');
subplot(4,1,4); plot(s);
title('quadrature phase shift keying');
xlabel('time');
ylabel('amplitude');

SIMULATION WAVEFORM
Program for QAM:
clc;
clear all;
M=16;
k=log2(M);
n=3*1e5;
nsamp=8;
X=randint(n,1);
xsym = bi2de(reshape(X,k,length(X)/k).','left-msb');
Y_qam= modulate([Link](M),xsym);
Ytx_qam = Y_qam;
EbNo=30;
SNR=EbNo+10*log10(k)-10*log10(nsamp);
Ynoisy_qam = awgn(Ytx_qam,SNR,'measured');
Yrx_qam = Ynoisy_qam;
h1=scatterplot(Yrx_qam(1:nsamp*5e3),nsamp,0,'r.');
hold on;
scatterplot(Yrx_qam(1:5e3),1,0,'k*',h1);
title('constellation diagram 16 QAM');
legend('Received signal' ,'signal constellation');
axis([-5 5 -5 5]);
hold off;

QAM CONSTELLATION

RESULT:

Thus the DPSK ,QPSK and QAM generation and Detection schemes were simulated using MATLAB.
OUTPUT

Enter the Generator Matrix: [1 0 0 0 1 0 1; 0 1 0 0 1 1 1; 0 0 1 0 1 1 0; 0 0 0 1 0 1


1] g =

1 0 0 0 1 0 1
0 1 0 0 1 1 1
0 0 1 0 1 1 0
0 0 0 1 0 1 1

G=
The Order of Linear block Code for given Generator Matrix
is: n =7
k =4

The Possible Code words


are: c =

0 0 0 0 0 0 0
0 0 0 1 0 1 1
0 0 1 0 1 1 0
0 0 1 1 1 0 1
0 1 0 0 1 1 1
0 1 0 1 1 0 0
0 1 1 0 0 0 1
0 1 1 1 0 1 0

1 0 0 0 1 0 1
1 0 0 1 1 1 0
1 0 1 0 0 1 1
1 0 1 1 0 0 0
1 1 0 0 0 1 0
1 1 0 1 0 0 1
1 1 1 0 1 0 0
1 1 1 1 1 1 1
[Link](a)
DATE:
SIMULATION OF LINEAR BLOCK CODING SCHEME
AIM
To simulate and study the error control coding scheme of linear block code using MATLAB

ALGORITHM:

STEP 1: Give the generator matrix


STEP 2: Find the order of the linear block code for the given generator matrix STEP 3: Obtain
the possible code words
STEP 4: Find the minimum hamming
distance STEP 5: Give the received code
word
STEP 6: Calculate the syndrome vector and compare it with transpose of
hamming matrix.
STEP 7: Find the error bit position and display the corrected code word

MATLAB CODE:

% Input Generator Matrix


g=input('Enter The Generator Matrix:
') disp ('G = ')
disp ('The Order of Linear block Code for given Generator Matrix
is:') [n,k] = size(transpose(g))
fori =
1:2^k forj
= k:-1:1
ifrem(i-1,2^(-j+k+1))>=2^(-
j+k) u(i,j)=1;
else
u(i,j)=
0; end
en
d
en
d
disp('The Possible Codewords
are :') c = rem(u*g,2)
disp('The Minimum Hamming Distance dmin for given Block Code is=
') d_min = min(sum((c(2:2^k,:))'))
The Minimum Hamming Distance dmin for given Block Code is=

d_min = 3

Enter the Received Code Word:[1 0 0 0 1 0 0]


r=
1 0 0 0 1 0 0
Hammimg Code
ht =

1 0 1
1 1 1
1 1 0
0 1 1
1 0 0
0 1 0
0 0 1
Syndrome of a Given Code word is :
s=
0 0 1
The Error is in bit:
7
The Corrected Code word is:
1 0 0 0 1 0 1

% Code Word
r= input('Enter the Received Code Word:')
p= [g(:,n-k+2:n)];

h= [transpose(p),eye(n-k)]; disp('HammimgCode') ht =
transpose(h)
disp('Syndrome of a Given Codeword is :')
s= rem(r*ht,2)
fori = 1:1:size(ht)
if(ht(i,1:3)==s)
r(i) = 1-r(i);
break;
end
end
disp('The Error is in
bit:') disp(i);
disp('The Corrected Codeword is
:') disp(r);
RESULT:

Thus the error control coding scheme of linear block code is studied using MATLAB.
OUTPUT
COMPUTATION OF CODE VECTORS FOR A CYCLIC CODE
Msg=
1001
1010
1011

Code =
1101001
0111010
0001011

SYNDROME DECODING

Recd=
1011110

Syndrome=7(decimal), 1 1 1(binary)

Parmat=
1001011
0101110
0010111

Corrvect=
0000010

Correctedcode=
1011100
[Link](b)
DATE:
SIMULATION OF ERROR CONTROL USING CYCLICCODE
AIM

a. To generate parity check matrix & generator matrix for a (7,4) Hamming code.
b. To generate parity check matrix given generator polynomial g(x) = 1+x+x3.
c. To determine the code vectors.
d. To perform syndrome decoding

PROGRAM:
Generation of parity check matrix and generator matrix for a (7, 4) Hamming code.
[h,g,n,k] = hammgen(3);
Generation of parity check matrix for the generator polynomial g(x) = 1+x+x3.
h1 = hammgen(3,[1011]);
Computation of code vectors for a cyclic code
clc;
close all;
n=7;
k=4;
msg=[1 0 0 1; 1 0 1 0; 1 0 1 1];
code = encode(msg,n,k,'cyclic');
msg
code
Syndrome decoding
clc;
close all;
q=3;
n=2^q-1;
k=n-q;
parmat = hammgen(q); % produce parity-check matrix
trt = syndtable(parmat); % produce decoding table
recd = [1 0 1 1 1 1 0 ] %received vector
syndrome = rem(recd * parmat',2);
syndrome_de = bi2de(syndrome, 'left-msb'); %convert to decimal
disp(['Syndrome = ',num2str(syndrome_de),.....
' (decimal), ',num2str(syndrome),' (binary) ']);
corrvect = trt(1+syndrome_de, :);%correction vector
correctedcode= rem(corrvect+recd,2);
parmat
corrvect
correctedcode
RESULT:

Thus encoding and decoding of block codes are performed using MATLAB.
[Link]
DATE:
10. SIMULATION OF CONVOLUTIONAL CODING SCHEME

AIM

To simulate convolutional coding scheme using MATLAB Simulink tool.


APPARATUS REQUIRED:

PC with MATLAB Software

ALGORITHM:

Simulate the link by following these steps:


1. Generate binary data.
2. Encode the data with a rate 2/3 convolutional code.
3. Modulate the encoded data.
4. Pass the signal through an AWGN channel.
5. Demodulate the received signal.
6. Decode the demodulated signal by using a Viterbi decoder.
7. Collect the error statistics.

THEORY:

Generating Random Data


The Bernoulli Binary Generator block produces the information source for this simulation. The block
generates a frame of three random bits at each sample time. The Samples per frame parameter
determines the number of rows of the output frame.
Convolutional Encoding with Puncturing
The Convolutional Encoder block encodes the data from the Bernoulli Binary
Generator. This example uses the same code as described in Soft-Decision Decoding.
The puncture pattern is specified by the Puncture vector parameter in the mask. The puncture vector
is a binary column vector. A 1 indicates that the bit in the corresponding position of the input vector
is sent to the output vector, while a 0 indicates that the bit is removed.
For example, to create a rate 3/4 code from the rate 1/2, constraint length 7 convolutional code, the
optimal puncture vector is [1 1 0 1 1 0].' (where the .' after the vector indicates the transpose). Bits in
positions 1, 2, 4, and 5 are transmitted, while bits in positions 3 and 6 are removed. Now, for every 3
bits of input, the punctured code generates 4 bits of output (as opposed to the 6 bits produced before
puncturing). This makes the rate 3/4.
PROGRAM:
%close all
%clear all
%clc
tt=tic();
Single_frame_size=96;
Total_no_of_Frames=100;
scatter_plot_frame=48;
x=1;
si=1;
pilt=3+3j;
no_of_Carriers=64;
intial_test_snr=0;
snr_step_size=2;
final_test_snr=50;
BER=zeros(Total_no_of_Frames,floor((final_test_snr+snr_step_size)/snr_step_size));
no_of_pilots=4;
no_of_data_without_carriers=(Single_frame_size*2/no_of_pilots)+ no_of_pilots;
cyclic_extension=16;
choice_snr=14;
t_data=floor(2*rand(1,Single_frame_size*Total_no_of_Frames));
t_rxed_data=zeros(2,Single_frame_size*Total_no_of_Frames);
ber=zeros(1,26);
BER=BER';
for col=1:26;
ber(1,col)=ber(1,col)+sum(BER(col,:));
end
ber=ber./(Single_frame_size*Total_no_of_Frames*log2(1000));
figure;
space=10;
t=1:space:Single_frame_size*Total_no_of_Frames;
subplot(211);
y=[t_rxed_data(1,1:space:Single_frame_size*Total_no_of_Frames);t_data(1:space:Single_frame_size*Total_no_of_Frame
s)]';
h=stem(t,y);
set(h(1),'MarkerFaceColor','blue')
set(h(2),'MarkerFaceColor','red','Marker','square')
title('Blue-->Processed received Signal at SNR=50 & Red-->Original generated data');
subplot(212);
y=[t_rxed_data(2,1:space:Single_frame_size*Total_no_of_Frames);t_data(1:space:Single_frame_size*Total_no_of_Frame
s)]';
h=stem(t,y);
set(h(1),'MarkerFaceColor','blue')
set(h(2),'MarkerFaceColor','red','Marker','square')
title_print=strcat('Blue-->Processed received Signal of SNR=',choice_snr,'& Red-->original generated data');
title(title_print);
figure;
i=0:2:16;
theoryBer=(1/4)*3/2*erfc(sqrt(4*0.1*(10.^(i/10))));
semilogy(i,theoryBer,'bs-','LineWidth',2);
hold on
semilogy(i,ber(1:9),'mx-','LineWidth',2);
title('BER vs SNR');
ylabel('Normalised BER');
xlabel('SNR(dB)');
grid on
legend('theory','simulation');
toc(tt);
OUTPUT:

RESULT:

Thus the convolution coding scheme was simulated using MATLAB Simulink tool.
Exp. No.: 14

Date:

LINE CODING AND DECODING TECHNIQUES

AIM:

1. To study the different line coding techniques with the communication trainer kit
.

APPARATUS REQUIRED:

2. Communication trainer kit.


3. Patch cords.
4. DSO/CRO

THEORY:
Line coding refers to the process of representing the bit stream (1’s and 0’s) in the form of
voltage or current variations optimally tuned for the specific properties of the physical channel
being used. The selection of a proper line code can help in so many ways: One possibility is to
aid in clock recovery at the receiver.
Some common types of line encoding in common-use nowadays are unipolar, polar,
bipolar, Manchester and Duobinary encoding. These codes are explained here:

Unipolar (Unipolar NRZ and Unipolar RZ):


Unipolar is the simplest line coding scheme possible. It has the advantage of being
compatible with TTL logic. Unipolar coding uses a positive rectangular pulse p(t) to represent
binary 1, and the absence of a pulse (i.e., zero voltage) to represent a binary 0. Two possibilities
for the pulse p(t) exist3: Non-Return-to-Zero (NRZ) rectangular pulse and Return-to-Zero (RZ)
rectangular pulse. The difference between Unipolar NRZ and Unipolar RZ codes is that the
rectangular pulse in NRZ stays at a positive value (e.g., +5V) for the full duration of the logic 1
bit, while the plus in RZ drops from +5V to 0V in the middle of the bit time.
A drawback of unipolar (RZ and NRZ) is that its average value is not zero, which means
it creates a significant DC-component at the receiver (see the impulse at zero frequency in the
corresponding power spectral density (PSD) of this line code.
MANCHESTER ENCODING:

In Manchester code each bit of data is signified by at least one transition. Manchester
encoding is therefore considered to be self-clocking, which means that accurate clock recovery
from a data stream is possible. In addition, the DC component of the encoded signal is zero.
Although transitions allow the signal to be self-clocking, it carries significant overhead as there
is a need for essentially twice the bandwidth of a simple NRZ or NRZI encoding
Ø Unipolar most of signal power is centered around origin and there is waste of power
due to DC component that is present.
Ø Polar format most of signal power is centered around origin and they are
simple to implement.
Ø Bipolar format does not have DC component and does not demand more bandwidth,
but power requirement is double than other formats.
Ø Manchester format does not have DC component but provides proper clocking.

PROCEDURE:

1. Connect the PRBS (test point P5) to various line coding formats. Obtain the coded
output as per the requirement.
2. Connect coded signal test point to corresponding decoding test point as inputs.
3. Set the SW1 as per the requirement.
4. Set the potentiometer P1 in minimum position.
5. Switch ON the power supply. Press the switch SW2 once.
6. Display the encoded signal and decoded signal on the cro
FUNCTIONAL BLOCK DIAGRAM:
TABULAR COLUMN:

Name of the Amplitude Time period Frequency


[Link] Signal Volt Sec Hz

MODEL GRAPH:
RESULT:

Thus the line coding and decoding techniques was studied.

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